Rate Control State-of-the-art Survey

The majority of Internet traffic use Transmission Control Protocol (TCP) as the transport level protocol. It provides a reliable ordered byte stream for the applications. However, applications such as live video streaming place an emphasis on timeliness over reliability. Also a smooth sending rate can be desirable over sharp changes in the sending rate. For these applications TCP is not necessarily suitable. Rate control attempts to address the demands of these applications. An important design feature in all rate control mechanisms is TCP friendliness. We should not negatively impact TCP performance since it is still the dominant protocol. Rate Control mechanisms are classified into two different mechanisms: window-based mechanisms and rate-based mechanisms. Window-based mechanisms increase their sending rate after a successful transfer of a window of packets similar to TCP. They typically decrease their sending rate sharply after a packet loss. Rate-based solutions control their sending rate in some other way. A large subset of rate-based solutions are called equation-based solutions. Equation-based solutions have a control equation which provides an allowed sending rate. Typically these rate-based solutions react slower to both packet losses and increases in available bandwidth making their sending rate smoother than that of window-based solutions. This report contains a survey of rate control mechanisms and a discussion of their relative strengths and weaknesses. A section is dedicated to a discussion on the enhancements in wireless environments. Another topic in the report is bandwidth estimation. Bandwidth estimation is divided into capacity estimation and available bandwidth estimation. We describe techniques that enable the calculation of a fair sending rate that can be used to create novel rate control mechanisms.Peer reviewe

Equation-Based Congestion Control for Unicast and Multicast Data Streams

We believe that the emergence of congestion control mechanisms for relatively-smooth congestion control for unicast and multicast traffic can play a key role in preventing the degradation of end-to-end congestion control in the public Internet, by providing a viable alternative for multimedia flows that would otherwise be tempted to avoid end-to-end congestion control altogether. The design of good congestion control mechanisms is a hard problem, even more so for multicast environments where scalability issues are much more of a concern than for unicast. In this dissertation, equation-based congestion control is presented as an alternative form of congestion control to the well-known TCP protocol. We focus on areas of equation-based congestion control which were not yet well understood and for which no adequate solutions existed. Starting from a unicast congestion control mechanism which in contrast to TCP provides smooth rate changes, we extend equation-based congestion control in several ways. We investigate how it can work together with applications which can only operate in a very limited region of available bandwidth and whose rate can thus not be adapted to the network conditions in the usual way. Such a congestion control mechanism can also complement conventional equation-based congestion control in regimes where available bandwidth is too low for further rate reduction. When extending unicast congestion control to multicast, it is of paramount importance to ensure that changes in the network conditions anywhere in the multicast tree are reported back to the sender as quickly as possible to allow the sender to adjust the rate accordingly. A scalable feedback mechanism that allows timely congestion feedback in the face of potentially very large receiver sets is one of the contributions of this dissertation. But also other components of a congestion control protocol, such as the rate increase/decrease policy or the slow-start mechanism, need to be adjusted to be able to use them in a multicast environment. Our resulting multicast congestion control protocol was implemented in a simulation environment for extensive protocol testing and turned into a library for the use in real-world applications. In addition, a simple video transmission tool was built for test purposes that uses this congestion control library

Enhanced transport protocols for real time and streaming applications on wireless links

Real time communications have, in the last decade, become a highly relevant component of Internet applications and services, with both interactive communications and streamed content being used in developed and developing countries alike. Due to the proliferation of mobile devices, wireless media is becoming the means of transmitting a large part of this increasingly important real time communications traffic. Wireless has also become an important technology in developing countries, with satellite communications being increasingly deployed for traffic backhaul and ubiquitous connection to the Internet. A number of issues need to be addressed in order to have an acceptable service quality for real time communications in wireless environments. In addition to this, the availability of multiple wireless interfaces on mobile devices presents an opportunity to improve and further exacerbates the issues already present on single wireless links. Therefore in this thesis, we consider improvements to transport protocols for real time communications and streaming services to address these problems and we provide the following contributions. To deal with wireless link issues of errors and delay, we propose two enhancements. First, an improvement technique for Datagram Congestion Control Protocol CCID4 for long delay wireless (e.g. satellite) links, demonstrating significant performance improvements for Voice over IP applications. To deal with link errors, we have proposed, implemented and evaluated an erasure coding based packet error correction approach for Concurrent Multipath Transfer extension of Stream Control Transport Protocol data transport over multiple wireless paths. We have identified packet reordering as a major cause of performance degradation in both single and multi-path transport protocols for real time communications and media streaming. We have proposed a dynamically resizable buffer based solution to mitigate this problem within the DCCP protocol. For improving the performance of multi-path transport protocols over dissimilar network paths, we have proposed a delay aware packet scheduling scheme, which significantly improves the performance of multimedia and bulk data transfer with CMT-SCTP in heterogeneous multi-path network scenarios. Finally, we have developed a tool for online streaming video quality evaluation experiments, comprising a real-time cross-layer video streaming technique implemented within an open-source H.264 video encoder tool called x264

Adaptive filtering of MPEG system streams in IP networks

Congestion and large differences in available link bandwidth create challenges for the design of applications that want to deliver high quality video over the Internet. We present an efficient adaptive filter for MPEG System streams that can be placed in the network (e.g., as an active service). This filter adjusts the bandwidth demands of an MPEG System stream to the available bandwidth without transcoding while maintaining synchronization between the streams embedded in the MPEG System. The filter is network-friendly: it is fair with respect to other (TCP) competing streams and it avoids generating bursty traffic. This paper presents the system architecture and an evaluation of our implementation in three different operating environments: a networking testbed in a laboratory environment, a home-user scenario (DSL line with 640Kbit/s), and a wide area network covering the Atlantic (server in Europe, client in the US). Moreover we examine the network-friendliness of the adaptation protocol and the relationship between the quality of the received continuous media and the protocol's aggressiveness. Our architecture is based on efficient MPEG System filtering to achieve high-quality video over best-effort network

Real-time data flow models and congestion management for wire and wireless IP networks

Includes abstract.Includes bibliographical references (leaves 103-111).In video streaming, network congestion compromises the video throughput performance and impairs its perceptual quality and may interrupt the display. Congestion control may take the form of rate adjustment through mechanisms by attempt to minimize the probability of congestion by adjusting the rate of the streaming video to match the available capacity of the network. This can be achieved either by adapting the quantization parameter of the video encoder or by varying the rate through a scalable video technique. This thesis proposes a congestion control protocol for streaming video where an interaction between the video source and the receiver is essential to monitor the network state. The protocol consists of adjusting the video transmission rate at the encoder whenever a change in the network conditions is observed and reported back to the sender

Toward a versatile transport protocol

Les travaux présentés dans cette thèse ont pour but d'améliorer la couche transport de l'architecture réseau de l'OSI. La couche transport est de nos jour dominée par l'utilisation de TCP et son contrôle de congestion. Récemment de nouveaux mécanismes de contrôle de congestion ont été proposés. Parmi eux TCP Friendly Rate Control (TFRC) semble être le plus abouti. Cependant, tout comme TCP, ce mécanisme ne prend pas en compte ni les évolutions du réseau ni les nouveaux besoins des applications. La première contribution de cette thèse consiste en une spécialisation de TFRC afin d'obtenir un protocole de transport avisé de la Qualité de Service (QdS) spécialement défini pour des réseaux à QdS offrant une garantie de bande passante. Ce protocole combine un mécanisme de contrôle de congestion orienté QdS qui prend en compte la réservation de bande passante au niveau réseau, avec un service de fiabilité totale afin de proposer un service similaire à TCP. Le résultat de cette composition constitue le premier protocole de transport adapté à des réseau à garantie de bande passante. En même temps que cette expansion de service au niveau réseau, de nouvelles technologies ont été proposées et déployées au niveau physique. Ces nouvelles technologies sont caractérisées par leur affranchissement de support filaire et la mobilité des systèmes terminaux. De plus, elles sont généralement déployées sur des entités où la puissance de calcul et la disponibilité mémoire sont inférieures à celles des ordinateurs personnels. La deuxième contribution de cette thèse est la proposition d'une adaptation de TFRC à ces entités via la proposition d'une version allégée du récepteur. Cette version a été implémentée, évaluée quantitativement et ses nombreux avantages et contributions ont été démontrés par rapport à TFRC. Enfin, nous proposons une optimisation des implémentations actuelles de TFRC. Cette optimisation propose tout d'abord un nouvel algorithme pour l'initialisation du récepteur basé sur l'utilisation de l'algorithme de Newton. Nous proposons aussi l'introduction d'un outil nous permettant d'étudier plus en détails la manière dont est calculé le taux de perte du côté récepteur. ABSTRACT : This thesis presents three main contributions that aim to improve the transport layer of the current networking architecture. The transport layer is nowadays overruled by the use of TCP and its congestion control. Recently new congestion control mechanisms have been proposed. Among them, TCP Friendly Rate Control (TFRC) appears to be one of the most complete. Nevertheless this congestion control mechanism, as TCP, does not take into account either the evolution of the network in terms of Quality of Service and mobility or the evolution of the applications. The first contribution of this thesis is a specialisation TFRC congestion control to propose a QoS-aware Transport Protocol specifically designed to operate over QoS-enabled networks with bandwidth guarantee mechanisms. This protocol combines a QoS-aware congestion control, which takes into account networklevel bandwidth reservations, with full reliability in order mechanism to provide a transport service similar to TCP. As a result, we obtain the guaranteed throughput at the application level where TCP fails. This protocol is the first transport protocol compliant with bandwidth guaranteed networks. At the same time the set of network services expands, new technologies have been proposed and deployed at the physical layer. These new technologies are mainly characterised by communications done without wire constraint and the mobility of the end-systems. Furthermore, these technologies are usually deployed on entities where the CPU power and memory storage are limited. The second contribution of this thesis is therefore to propose an adaptation of TFRC to these entities. This is accomplished with the proposition of a new sender-based version of TFRC. This version has been implemented, evaluated and its numerous contributions and advantages compare to usual TFRC version have been demonstrated. Finally, we proposed an optimisation of actual implementations of TFRC. This optimisation first consists in the proposition of an algorithm based on a numerical analysis of the equation used in TFRC and the use of the Newton's algorithm. We furthermore give a first step, with the introduction of a new framework for TFRC, in order to better understand TFRC behaviour and to optimise the computation of the packet loss rate according to loss probability distribution

Best effort measurement based congestion control

Abstract available: p.

Performance and TCP-friendliness of the SQRT congestion control protocol in lossy environments

In this letter, an accurate analytic model to predict the send rate of the SQRT binomial congestion control protocol is derived, under the assumption that the protocol operates in an environment where the packet loss rate is relatively high. Based on this model, a study of its TCP-friendliness is performed, remarking the nonnegligible effects on fairness that different queue managements have

Bioinspired metaheuristic algorithms for global optimization

This paper presents concise comparison study of newly developed bioinspired algorithms for global optimization problems. Three different metaheuristic techniques, namely Accelerated Particle Swarm Optimization (APSO), Firefly Algorithm (FA), and Grey Wolf Optimizer (GWO) are investigated and implemented in Matlab environment. These methods are compared on four unimodal and multimodal nonlinear functions in order to find global optimum values. Computational results indicate that GWO outperforms other intelligent techniques, and that all aforementioned algorithms can be successfully used for optimization of continuous functions